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Westerlund 3 Internet-Draft Ericsson 4 Obsoletes: 5117 (if approved) S. Wenger 5 Intended status: Informational Vidyo 6 Expires: February 8, 2015 August 7, 2014 8 RTP Topologies 9 draft-ietf-avtcore-rtp-topologies-update-03 11 Abstract 13 This document discusses point to point and multi-endpoint topologies 14 used in Real-time Transport Protocol (RTP)-based environments. In 15 particular, centralized topologies commonly employed in the video 16 conferencing industry are mapped to the RTP terminology. 18 This document is updated with additional topologies and is intended 19 to replace RFC 5117. 21 Status of This Memo 23 This Internet-Draft is submitted in full conformance with the 24 provisions of BCP 78 and BCP 79. 26 Internet-Drafts are working documents of the Internet Engineering 27 Task Force (IETF). Note that other groups may also distribute 28 working documents as Internet-Drafts. The list of current Internet- 29 Drafts is at http://datatracker.ietf.org/drafts/current/. 31 Internet-Drafts are draft documents valid for a maximum of six months 32 and may be updated, replaced, or obsoleted by other documents at any 33 time. It is inappropriate to use Internet-Drafts as reference 34 material or to cite them other than as "work in progress." 36 This Internet-Draft will expire on February 8, 2015. 38 Copyright Notice 40 Copyright (c) 2014 IETF Trust and the persons identified as the 41 document authors. All rights reserved. 43 This document is subject to BCP 78 and the IETF Trust's Legal 44 Provisions Relating to IETF Documents 45 (http://trustee.ietf.org/license-info) in effect on the date of 46 publication of this document. Please review these documents 47 carefully, as they describe your rights and restrictions with respect 48 to this document. Code Components extracted from this document must 49 include Simplified BSD License text as described in Section 4.e of 50 the Trust Legal Provisions and are provided without warranty as 51 described in the Simplified BSD License. 53 Table of Contents 55 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 56 2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 3 57 2.1. Glossary . . . . . . . . . . . . . . . . . . . . . . . . 3 58 2.2. Definitions related to RTP grouping taxonomy . . . . . . 4 59 3. Topologies . . . . . . . . . . . . . . . . . . . . . . . . . 5 60 3.1. Point to Point . . . . . . . . . . . . . . . . . . . . . 5 61 3.2. Point to Point via Middlebox . . . . . . . . . . . . . . 6 62 3.2.1. Translators . . . . . . . . . . . . . . . . . . . . . 6 63 3.2.2. Back to Back RTP sessions . . . . . . . . . . . . . . 10 64 3.3. Point to Multipoint Using Multicast . . . . . . . . . . . 11 65 3.3.1. Any Source Multicast (ASM) . . . . . . . . . . . . . 11 66 3.3.2. Source Specific Multicast (SSM) . . . . . . . . . . . 12 67 3.3.3. SSM with Local Unicast Resources . . . . . . . . . . 14 68 3.4. Point to Multipoint Using Mesh . . . . . . . . . . . . . 16 69 3.5. Point to Multipoint Using the RFC 3550 Translator . . . . 19 70 3.5.1. Relay - Transport Translator . . . . . . . . . . . . 19 71 3.5.2. Media Translator . . . . . . . . . . . . . . . . . . 20 72 3.6. Point to Multipoint Using the RFC 3550 Mixer Model . . . 21 73 3.6.1. Media Mixing Mixer . . . . . . . . . . . . . . . . . 23 74 3.6.2. Media Switching . . . . . . . . . . . . . . . . . . . 26 75 3.7. Selective Forwarding Middlebox . . . . . . . . . . . . . 28 76 3.8. Point to Multipoint Using Video Switching MCUs . . . . . 31 77 3.9. Point to Multipoint Using RTCP-Terminating MCU . . . . . 33 78 3.10. Split Component Terminal . . . . . . . . . . . . . . . . 34 79 3.11. Non-Symmetric Mixer/Translators . . . . . . . . . . . . . 36 80 3.12. Combining Topologies . . . . . . . . . . . . . . . . . . 36 81 4. Comparing Topologies . . . . . . . . . . . . . . . . . . . . 37 82 4.1. Topology Properties . . . . . . . . . . . . . . . . . . . 37 83 4.1.1. All to All Media Transmission . . . . . . . . . . . . 37 84 4.1.2. Transport or Media Interoperability . . . . . . . . . 38 85 4.1.3. Per Domain Bit-Rate Adaptation . . . . . . . . . . . 38 86 4.1.4. Aggregation of Media . . . . . . . . . . . . . . . . 39 87 4.1.5. View of All Session Participants . . . . . . . . . . 39 88 4.1.6. Loop Detection . . . . . . . . . . . . . . . . . . . 39 89 4.2. Comparison of Topologies . . . . . . . . . . . . . . . . 40 90 5. Security Considerations . . . . . . . . . . . . . . . . . . . 40 91 6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 42 92 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 43 93 8. References . . . . . . . . . . . . . . . . . . . . . . . . . 43 94 8.1. Normative References . . . . . . . . . . . . . . . . . . 43 95 8.2. Informative References . . . . . . . . . . . . . . . . . 43 96 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 44 98 1. Introduction 100 Real-time Transport Protocol (RTP) [RFC3550] topologies describe 101 methods for interconnecting RTP entities and their processing 102 behavior of RTP and RTCP. This document tries to address past and 103 existing confusion, especially with respect to terms not defined in 104 RTP but in common use in the conversational communication industry, 105 such as the Multipoint Control Unit or MCU. 107 When the Audio-Visual Profile with Feedback (AVPF) [RFC4585] was 108 developed the main emphasis lay in the efficient support of point to 109 point and small multipoint scenarios without centralized multipoint 110 control. In practice, however, most multipoint conferences operate 111 utilizing centralized units referred to as MCUs. MCUs may implement 112 Mixer or Translator functionality (in RTP [RFC3550] terminology), and 113 signalling support. They may also contain additional application 114 layer functionality. This document focuses on the media transport 115 aspects of the MCU that can be realized using RTP, as discussed 116 below. Further considered are the properties of Mixers and 117 Translators, and how some types of deployed MCUs deviate from these 118 properties. 120 This document also codifies new multipoint architectures that have 121 recently been introduced and which were not anticipated in RFC 5117. 122 These architectures use scalable video coding and simulcasting, and 123 their associated centralized units are referred to as Selective 124 Forwarding Units (SFU). This codification provides a common 125 information basis for future discussion and specification work. 127 The document's attempt to clarify and explain sections of the Real- 128 time Transport Protocol (RTP) spec [RFC3550] is informal. It is not 129 intended to update or change what is normatively specified within RFC 130 3550. 132 2. Definitions 134 2.1. Glossary 136 ASM: Any Source Multicast 138 AVPF: The Extended RTP Profile for RTCP-based Feedback 140 CSRC: Contributing Source 142 Link: The data transport to the next IP hop 144 Middlebox: A device that is on the Path that media travel between 145 two Endpoints 147 MCU: Multipoint Control Unit 149 Path: The concatenation of multiple links, resulting in an end-to- 150 end data transfer. 152 PtM: Point to Multipoint 154 PtP: Point to Point 156 SFU: Selective Forwarding Unit 158 SSM: Source-Specific Multicast 160 SSRC: Synchronization Source 162 2.2. Definitions related to RTP grouping taxonomy 164 [Note to RFC editor: The following definitions have been taken from 165 draft-ietf-avtext-rtp-grouping-taxonomy-02 (taxonomy draft 166 henceforth). It is avtcore working group agreement to not delay the 167 publication of the topologies-update document through a dependency to 168 the taxonomy draft. If, however, the taxonomy draft and this draft 169 are in your work queue at the same time and there would be no 170 significant additional delay (through your schedule, normative 171 reference citations, or similar) in publishing both documents roughly 172 in parallel, it would be preferable to replace the definition 173 language with something like "as in [RFC YYYY]" where YYYY would be 174 the RFC number of the published taxonomy draft.] 176 The following definitions have been taken from draft-ietf-avtext-rtp- 177 grouping-taxonomy-02, and are used in capitalized form throughout the 178 document. 180 Communication Session: A Communication Session is an association 181 among group of participants communicating with each other via a 182 set of Multimedia Sessions. 184 End Point: A single addressable entity sending or receiving RTP 185 packets. It may be decomposed into several functional blocks, but 186 as long as it behaves as a single RTP stack entity it is 187 classified as a single "End Point". 189 Media Source: A Media Source is the logical source of a reference 190 clock synchronized, time progressing, digital media stream, called 191 a Source Stream. 193 Multimedia Session: A multimedia session is an association among a 194 group of participants engaged in the communication via one or more 195 RTP Sessions. 197 3. Topologies 199 This subsection defines several topologies that are relevant for 200 codec control but also RTP usage in other contexts. The section 201 starts with point to point cases, with or without middleboxes. Then 202 follows a number of different methods for establishing point to 203 multipoint communication. These are structured around the most 204 fundamental enabler, i.e., multicast, a mesh of connections, 205 translators, mixers and finally MCUs and SFUs. The section ends by 206 discussing de-composited terminals, asymmetric middlebox behaviors 207 and combining topologies. 209 The topologies may be referenced in other documents by a shortcut 210 name, indicated by the prefix "Topo-". 212 For each of the RTP-defined topologies, we discuss how RTP, RTCP, and 213 the carried media are handled. With respect to RTCP, we also discuss 214 the handling of RTCP feedback messages as defined in [RFC4585] and 215 [RFC5104]. 217 3.1. Point to Point 219 Shortcut name: Topo-Point-to-Point 221 The Point to Point (PtP) topology (Figure 1) consists of two End 222 Points, communicating using unicast. Both RTP and RTCP traffic are 223 conveyed endpoint-to-endpoint, using unicast traffic only (even if, 224 in exotic cases, this unicast traffic happens to be conveyed over an 225 IP-multicast address). 227 +---+ +---+ 228 | A |<------->| B | 229 +---+ +---+ 231 Figure 1: Point to Point 233 The main property of this topology is that A sends to B, and only B, 234 while B sends to A, and only A. This avoids all complexities of 235 handling multiple End Points and combining the requirements stemming 236 from them. Note that an End Point can still use multiple RTP 237 Synchronization Sources (SSRCs) in an RTP session. The number of RTP 238 sessions in use between A and B can also be of any number, subject 239 only to system level limitations like the number range of ports. 241 RTCP feedback messages for the indicated SSRCs are communicated 242 directly between the End Points. Therefore, this topology poses 243 minimal (if any) issues for any feedback messages. For RTP sessions 244 which use multiple SSRC per End Point it can be relevant to implement 245 support for cross-reporting suppression as defined in "Sending 246 Multiple Media Streams in a Single RTP Session" 247 [I-D.ietf-avtcore-rtp-multi-stream-optimisation]. 249 3.2. Point to Point via Middlebox 251 This section discusses cases where two End Points communicate but 252 have one or more middleboxes involved in the RTP session. 254 3.2.1. Translators 256 Shortcut name: Topo-PtP-Translator 258 Two main categories of Translators can be distinguished; Transport 259 Translators and Media translators. Both Translator types share 260 common attributes that separate them from Mixers. For each RTP 261 stream that the Translator receives, it generates an individual RTP 262 stream in the other domain. A translator keeps the SSRC for an RTP 263 stream across the translation, whereas a Mixer can select a single 264 RTP stream from multiple received RTP streams (in cases like audio/ 265 video switching), or send out an RTP stream composed of multiple 266 mixed media received in multiple RTP streams (in cases like audio 267 mixing or video tiling), but always under its own SSRC, possibly 268 using the CSRC field to indicate the source(s) of the content. 269 Mixers are more common in point to multipoint cases than in PtP. The 270 reason is that in PtP use cases the primary focus of a middlebox is 271 enabling interoperability, between otherwise non-interoperable End 272 Points, such as transcoding to a codec the receiver supports, which 273 can be done by a media translator. 275 As specified in Section 7.1 of [RFC3550], the SSRC space is common 276 for all participants in the RTP session, independent of on which side 277 of the Translator the session resides. Therefore, it is the 278 responsibility of the End Points (as the RTP session participants) to 279 run SSRC collision detection, and the SSRC is thus a field the 280 Translator cannot change. Any SDES information associated with a 281 SSRC or CSRC also needs to be forwarded between the domains for any 282 SSRC/CSRC used in the different domains. 284 A Translator commonly does not use an SSRC of its own, and is not 285 visible as an active participant in the RTP session. One reason to 286 have its own SSRC is when a Translator acts as a quality monitor that 287 sends RTCP reports and therefore is required to have an SSRC. 288 Another example is the case when a Translator is prepared to use RTCP 289 feedback messages. This may, for example, occur in a translator 290 configured to detect packet loss of important video packets and wants 291 to trigger repair by the media sending End Point, by sending feedback 292 messages. While such feedback could use the SSRC of the target for 293 the translator (the receiving End Point), this in turn would require 294 translation of the targets RTCP reports to make them consistent. It 295 may be simpler to expose an additional SSRC in the session. The only 296 concern is End Points failing to support the full RTP specification 297 may have issues with multiple SSRCs reporting on the RTP streams sent 298 by that End Point, as this use case may be viewed as excotic by 299 implementers. 301 In general, a Translator implementation should consider which RTCP 302 feedback messages or codec-control messages it needs to understand in 303 relation to the functionality of the Translator itself. This is 304 completely in line with the requirement to also translate RTCP 305 messages between the domains. 307 3.2.1.1. Transport Relay/Anchoring 309 There exist a number of different types of middleboxes that might be 310 inserted between two End Points on the transport level, e.g., to 311 perform changes on the IP/UDP headers, and are, therefore, basic 312 transport translators. These middleboxes come in many variations 313 including NAT [RFC3022] traversal by pinning the media path to a 314 public address domain relay, network topologies where the RTP stream 315 is required to pass a particular point for audit by employing 316 relaying, or preserving privacy by hiding each peer's transport 317 addresses to the other party. Other protocols or functionalities 318 that provide this behavior are TURN [RFC5766] servers, Session Border 319 Gateways and Media Processing Nodes with media anchoring 320 functionalities. 322 +---+ +---+ +---+ 323 | A |<------>| T |<------->| B | 324 +---+ +---+ +---+ 326 Figure 2: Point to Point with Translator 328 A common element in these functions is that they are normally 329 transparent at the RTP level, i.e., they perform no changes on any 330 RTP or RTCP packet fields and only affect the lower layers. They may 331 affect, however, the path the RTP and RTCP packets are routed between 332 the End Points in the RTP session, and thereby indirectly affect the 333 RTP session. For this reason, one could believe that transport 334 translator-type middleboxes do not need to be included in this 335 document. This topology, however, can raise additional requirements 336 in the RTP implementation and its interactions with the signalling 337 solution. Both in signalling and in certain RTCP fields, network 338 addresses other than those of the relay can occur since B has a 339 different network address than the relay (T). Implementations that 340 cannot support this will also not work correctly when End Points are 341 subject to NAT. 343 The transport relay implementations also have to take into account 344 security considerations. In particular, source address filtering of 345 incoming packets is usually important in relays, to prevent attackers 346 to inject traffic into a session, which one peer may, in the absence 347 fo adequate security in the relay, think it comes from the other 348 peer. 350 3.2.1.2. Transport Translator 352 Transport Translators (Topo-Trn-Translator) do not modify the RTP 353 stream itself, but are concerned with transport parameters. 354 Transport parameters, in the sense of this section, comprise the 355 transport addresses (to bridge different domains such unicast to 356 multicast) and the media packetization to allow other transport 357 protocols to be interconnected to a session (in gateways). 359 Translators that bridge between different protocol worlds need to be 360 concerned about the mapping of the SSRC/CSRC (Contributing Source) 361 concept to the non-RTP protocol. When designing a Translator to a 362 non-RTP-based media transport, an important consideration is how to 363 handle different sources and their identities. This problem space is 364 not discussed henceforth. 366 Of the transport Translators, this memo is primarily interested in 367 those that use RTP on both sides, and this is assumed henceforth. 369 The most basic transport translators that operate below the RTP level 370 were already discussed in Section 3.2.1.1. 372 3.2.1.3. Media Translator 374 Media Translators (Topo-Media-Translator) modify the media inside the 375 RTP stream. This process is commonly known as transcoding. The 376 modification of the media can be as small as removing parts of the 377 stream, and it can go all the way to a full decoding and re-encoding 378 (down to the sample level or equivalent) utilizing a different media 379 codec. Media Translators are commonly used to connect End Points 380 without a common interoperability point in the media encoding. 382 Stand-alone Media Translators are rare. Most commonly, a combination 383 of Transport and Media Translator is used to translate both the media 384 and the transport aspects of the RTP stream carrying the media 385 between two transport domains. 387 When media translation occurs, the Translator's task regarding 388 handling of RTCP traffic becomes substantially more complex. In this 389 case, the Translator needs to rewrite End Point B's RTCP Receiver 390 Report before forwarding them to End Point A. The rewriting is 391 needed as the RTP stream received by B is not the same RTP stream as 392 the other participants receive. For example, the number of packets 393 transmitted to B may be lower than what A sends, due to the different 394 media format and data rate. Therefore, if the Receiver Reports were 395 forwarded without changes, the extended highest sequence number would 396 indicate that B were substantially behind in reception, while most 397 likely it would not be. Therefore, the Translator must translate 398 that number to a corresponding sequence number for the stream the 399 Translator received. Similar requirements exists for most other 400 fields in the RTCP Receiver Reports. 402 A media Translator may in some cases act on behalf of the "real" 403 source (the End Point originally sending the media to the Translator) 404 and respond to RTCP feedback messages. This may occur, for example, 405 when a receiving End Point requests a bandwidth reduction, and the 406 media Translator has not detected any congestion or other reasons for 407 bandwidth reduction between the sending End Point and itself. In 408 that case, it is sensible that the media Translator reacts to codec 409 control messages itself, for example, by transcoding to a lower media 410 rate. 412 A variant of translator behaviour worth pointing out is the one 413 depicted in Figure 3 of an End Point A sending a RTP stream 414 containing media (only) to B. On the path there is a device T that 415 on A's behalf manipulates the RTP streams. One common example is 416 that T adds a second RTP stream containing Forward Error Correction 417 (FEC) information in order to protect A's (non FEC-protected) RTP 418 stream. In this case, T needs to semantically bind the new FEC RTP 419 stream to A's media-carrying RTP stream, for example by using the 420 same CNAME as A. 422 +------+ +------+ +------+ 423 | | | | | | 424 | A |------->| T |-------->| B | 425 | | | |---FEC-->| | 426 +------+ +------+ +------+ 428 Figure 3: Media Translator adding FEC 430 there may also be cases where information is added into the original 431 RTP stream, while leaving most or all of the original RTP packets 432 intact (with the exception of certain RTP header fields, such as the 433 sequence number). One example is the injection of meta-data into the 434 RTP stream, carried in their own RTP packets. 436 Similarly, a Media Translator can sometimes remove information from 437 the RTP stream, while otherwise leaving teh remaining RTP packets 438 unchanged (again with the exception of certain RTP header fields). 440 Either type of functionality where T manipulates the RTP stream, or 441 adds an accompanying RTP stream, on behalf of A is also covered under 442 the media translator definition. 444 3.2.2. Back to Back RTP sessions 446 There exist middleboxes that interconnect two End Points A and B 447 through themselves (MB), but not by being part of a common RTP 448 session. They establish instead two different RTP sessions, one 449 between A and the middlebox and another between the middlebox and B. 450 This topology is called Topo-Back-To-Back 452 |<--Session A-->| |<--Session B-->| 453 +------+ +------+ +------+ 454 | A |------->| MB |-------->| B | 455 +------+ +------+ +------+ 457 Figure 4: Back-to-back RTP sessions through Middlebox 459 The middlebox acts as an application-level gateway and bridges the 460 two RTP sessions. This bridging can be as basic as forwarding the 461 RTP payloads between the sessions, or more complex including media 462 transcoding. The difference of this topology relative to the single 463 RTP session context is the handling of the SSRCs and the other 464 session-related identifiers, such as CNAMEs. With two different RTP 465 sessions these can be freely changed and it becomes the middlebox's 466 respnsibility to maintain the correct relations. 468 The signalling or other above-RTP level functionalities referencing 469 RTP streams may be what is most impacted by using two RTP sessions 470 and changing identifiers. The structure with two RTP sessions also 471 puts a congestion control requirement on the middlebox, because it 472 becomes fully responsible for the media stream it sources into each 473 of the sessions. 475 Adherence to congestion control can be solved locally on each of the 476 two segments, or by bridging statistics from the receiving End Point 477 through the middlebox to the sending End Point. From an 478 implementation point, however, the latter requires dealing with a 479 number of inconsistencies. First, packet loss must be detected for 480 an RTP stream sent from A to the middlebox, and that loss must be 481 reported through a skipped sequence number in the RTP stream from the 482 middlebox to B. This coupling and the resulting inconsistencies are 483 conceptually easier to handle when considering the two RTP streams as 484 belonging to a single RTP session. 486 3.3. Point to Multipoint Using Multicast 488 Multicast is an IP layer functionality that is available in some 489 networks. Two main flavors can be distinguished: Any Source 490 Multicast (ASM) [RFC1112] where any multicast group participant can 491 send to the group address and expect the packet to reach all group 492 participants; and Source Specific Multicast (SSM) [RFC3569], where 493 only a particular IP host sends to the multicast group. Both these 494 models are discussed below in their respective sections. 496 3.3.1. Any Source Multicast (ASM) 498 Shortcut name: Topo-ASM (was Topo-Multicast) 500 +-----+ 501 +---+ / \ +---+ 502 | A |----/ \---| B | 503 +---+ / Multi- \ +---+ 504 + Cast + 505 +---+ \ Network / +---+ 506 | C |----\ /---| D | 507 +---+ \ / +---+ 508 +-----+ 510 Figure 5: Point to Multipoint Using Multicast 512 Point to Multipoint (PtM) is defined here as using a multicast 513 topology as a transmission model, in which traffic from any multicast 514 group participant reaches all the other multicast group participants, 515 except for cases such as: 517 o packet loss, or 519 o when a multicast group participant does not wish to receive the 520 traffic for a specific multicast group and, therefore, has not 521 subscribed to the IP multicast group in question. This scenario 522 can occur, for example, where a multimedia session is distributed 523 using two or more multicast groups and a multicast group 524 participant is subscribed only to a subset of these sessions. 526 In the above context, "traffic" encompasses both RTP and RTCP 527 traffic. The number of multicast group participants can vary between 528 one and many, as RTP and RTCP scale to very large multicast groups 529 (the theoretical limit of the number of participants in a single RTP 530 session is in the range of billions). The above can be realized 531 using Any Source Multicast (ASM). 533 For feedback usage, it is useful to define a "small multicast group" 534 as a group where the number of multicast group participants is so low 535 (and other factors such as the connectivity is so good) that it 536 allows the participants to use early or immediate feedback, as 537 defined in AVPF [RFC4585]. Even when the environment would allow for 538 the use of a small multicast group, some applications may still want 539 to use the more limited options for RTCP feedback available to large 540 multicast groups, for example when there is a likelihood that the 541 threshold of the small multicast group (in terms of multicast group 542 participants) may be exceeded during the lifetime of a session. 544 RTCP feedback messages in multicast reach, like media data, every 545 subscriber (subject to packet losses and multicast group 546 subscription). Therefore, the feedback suppression mechanism 547 discussed in [RFC4585] is typically required. Each individual End 548 Point that is a multicast group participant needs to process every 549 feedback message it receives, not only to determine if it is affected 550 or if the feedback message applies only to some other End Point, but 551 also to derive timing restrictions for the sending of its own 552 feedback messages, if any. 554 3.3.2. Source Specific Multicast (SSM) 556 In Any Source Multicast, any of the multicast group participants can 557 send to all the other multicast group participants, by sending a 558 packet to the multicast group. In contrast, Source Specific 559 Multicast [RFC3569][RFC4607] refers to scenarios where only a single 560 source (Distribution Source) can send to the multicast group, 561 creating a topology that looks like the one below: 563 +--------+ +-----+ 564 |Media | | | Source-specific 565 |Sender 1|<----->| D S | Multicast 566 +--------+ | I O | +--+----------------> R(1) 567 | S U | | | | 568 +--------+ | T R | | +-----------> R(2) | 569 |Media |<----->| R C |->+ | : | | 570 |Sender 2| | I E | | +------> R(n-1) | | 571 +--------+ | B | | | | | | 572 : | U | +--+--> R(n) | | | 573 : | T +-| | | | | 574 : | I | |<---------+ | | | 575 +--------+ | O |F|<---------------+ | | 576 |Media | | N |T|<--------------------+ | 577 |Sender M|<----->| | |<-------------------------+ 578 +--------+ +-----+ RTCP Unicast 580 FT = Feedback Target 581 Transport from the Feedback Target to the Distribution 582 Source is via unicast or multicast RTCP if they are not 583 co-located. 585 Figure 6: Point to Multipoint using Source Specific Multicast 587 In the SSM topology (Figure 6) a number of RTP sending End Points 588 (RTP sources henceforth) (1 to M) are allowed to send media to the 589 SSM group. These sources send media to a dedicated distribution 590 source, which forwards the RTP streams to the multicast group on 591 behalf of the original RTP sources. The RTP streams reach the 592 receiving End Points (Receivers henceforth) (R(1) to R(n)). The 593 Receivers' RTCP messages cannot be sent to the multicast group, as 594 the SSM multicast group by definition has only a single IP sender. 595 To support RTCP, an RTP extension for SSM [RFC5760] was defined. It 596 uses unicast transmission to send RTCP from each of the receivers to 597 one or more Feedback Targets (FT). The feedback targets relay the 598 RTCP unmodified, or provide a summary of the participants RTCP 599 reports towards the whole group by forwarding the RTCP traffic to the 600 distribution source. Figure 6 only shows a single feedback target 601 integrated in the distribution source, but for scalability the FT can 602 be distributed and each instance can have responsibility for sub- 603 groups of the receivers. For summary reports, however, there 604 typically must be a single feedback target aggregating all the 605 summaries to a common message to the whole receiver group. 607 The RTP extension for SSM specifies how feedback (both reception 608 information and specific feedback events) are handled. The more 609 general problems associated with the use of multicast, where everyone 610 receives what the distribution source sends needs to be accounted 611 for. 613 Aforementioned situation results in common behavior for RTP 614 multicast: 616 1. Multicast applications often use a group of RTP sessions, not 617 one. Each End Point needs to be a member of most or all of these 618 RTP sessions in order to perform well. 620 2. Within each RTP session, the number of media sinks is likely to 621 be much larger than the number of RTP sources. 623 3. Multicast applications need signalling functions to identify the 624 relationships between RTP sessions. 626 4. Multicast applications need signalling functions to identify the 627 relationships between SSRCs in different RTP sessions. 629 All multicast configurations share a signalling requirement: all of 630 the End Points need to have the same RTP and payload type 631 configuration. Otherwise, End Point A could, for example, be using 632 payload type 97 to identify the video codec H.264, while End Point B 633 would identify it as MPEG-2, with unpredicatble but almost certainly 634 not visually pleasing results. 636 Security solutions for this type of group communications are also 637 challenging. First, the key-management and the security protocol 638 must support group communication. Source authentication becomes more 639 difficult and requires specialized solutions. For more discussion on 640 this please review Options for Securing RTP Sessions [RFC7201]. 642 3.3.3. SSM with Local Unicast Resources 644 [RFC6285] "Unicast-Based Rapid Acquisition of Multicast RTP Sessions" 645 results in additional extensions to SSM Topology. 647 ----------- -------------- 648 | |------------------------------------>| | 649 | |.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.->| | 650 | | | | 651 | Multicast | ---------------- | | 652 | Source | | Retransmission | | | 653 | |-------->| Server (RS) | | | 654 | |.-.-.-.->| | | | 655 | | | ------------ | | | 656 ----------- | | Feedback | |<.=.=.=.=.| | 657 | | Target (FT)| |<~~~~~~~~~| RTP Receiver | 658 PRIMARY MULTICAST | ------------ | | (RTP_Rx) | 659 RTP SESSION with | | | | 660 UNICAST FEEDBACK | | | | 661 | | | | 662 - - - - - - - - - - - |- - - - - - - - |- - - - - |- - - - - - - |- - 663 | | | | 664 UNICAST BURST | ------------ | | | 665 (or RETRANSMISSION) | | Burst/ | |<~~~~~~~~>| | 666 RTP SESSION | | Retrans. | |.........>| | 667 | |Source (BRS)| |<.=.=.=.=>| | 668 | ------------ | | | 669 | | | | 670 ---------------- -------------- 672 -------> Multicast RTP Stream 673 .-.-.-.> Multicast RTCP Stream 674 .=.=.=.> Unicast RTCP Reports 675 ~~~~~~~> Unicast RTCP Feedback Messages 676 .......> Unicast RTP Stream 678 Figure 7 680 The Rapid acquisition extension allows an End Point joining an SSM 681 multicast session to request media starting with the last sync-point 682 (from where media can be decoded without requiring context 683 established by the decoding of prior packets) to be sent at high 684 speed until such time where, after decoding of these burst-delivered 685 media packets, the correct media timing is established, i.e. media 686 packets are received within adequate buffer intervals for this 687 application. This is accomplished by first establishing a unicast 688 PtP RTP session between the Burst/Retransmission Source (BRS, 689 Figure 7) and the RTP Receiver. The unicast session is used to 690 transmit cached packets from the multicast group at higher then 691 normal speed in order to synchronize the receiver to the ongoing 692 multicast RTP stream. Once the RTP receiver and its decoder have 693 caught up with the multicast session's current delivery, the receiver 694 switches over to receiving directly from the multicast group. The 695 (still existing) PtP RTP session is, in many deployed applications, 696 be used as a repair channel, i.e., for RTP Retransmission traffic of 697 those packets that were not received from the multicast group. 699 3.4. Point to Multipoint Using Mesh 701 Shortcut name: Topo-Mesh 703 +---+ +---+ 704 | A |<---->| B | 705 +---+ +---+ 706 ^ ^ 707 \ / 708 \ / 709 v v 710 +---+ 711 | C | 712 +---+ 714 Figure 8: Point to Multi-Point using Mesh 716 Based on the RTP session definition, it is clearly possible to have a 717 joint RTP session involving three or more End Points over multiple 718 unicast transport flows, like the joint three End point session 719 depicted above. In this case, A needs to send its RTP streams and 720 RTCP packets to both B and C over their respective transport flows. 721 As long as all End Points do the same, everyone will have a joint 722 view of the RTP session. 724 This topology does not create any additional requirements beyond the 725 need to have multiple transport flows associated with a single RTP 726 session. Note that an End Point may use a single local port to 727 receive all these transport flows (in which case the sending port, IP 728 address, or SSRC can be used to demultiplex), or it might have 729 separate local reception ports for each of the End Points. 731 +-A--------------------+ 732 |+---+ | 733 ||CAM| | +-B-----------+ 734 |+---+ +-UDP1------| |-UDP1------+ | 735 | | | +-RTP1----| |-RTP1----+ | | 736 | V | | +-Video-| |-Video-+ | | | 737 |+----+ | | | |<----------------|BV1 | | | | 738 ||ENC |----+-+-+--->AV1|---------------->| | | | | 739 |+----+ | | +-------| |-------+ | | | 740 | | | +---------| |---------+ | | 741 | | +-----------| |-----------+ | 742 | | | +-------------+ 743 | | | 744 | | | +-C-----------+ 745 | | +-UDP2------| |-UDP2------+ | 746 | | | +-RTP1----| |-RTP1----+ | | 747 | | | | +-Video-| |-Video-+ | | | 748 | +-------+-+-+--->AV1|---------------->| | | | | 749 | | | | |<----------------|CV1 | | | | 750 | | | +-------| |-------+ | | | 751 | | +---------| |---------+ | | 752 | +-----------| |-----------+ | 753 +----------------------+ +-------------+ 755 Figure 9: An Multi-unicast Mesh with a joint RTP session 757 A joint RTP session from End Point A's perspective for the Mesh 758 depicted in Figure 8 with a joint RTP session have multiple transport 759 flows, here enumerated as UDP1 and UDP2. However, there is only one 760 RTP session (RTP1). The Media Source (CAM) is encoded and 761 transmitted over the SSRC (AV1) across both transport layers. 762 However, as this is a joint RTP session, the two streams must be the 763 same. Thus, an congestion control adaptation needed for the paths A 764 to B and A to C needs to use the most restricting path's properties. 766 An alternative structure for establishing the above topology is to 767 use independent RTP sessions between each pair of peers, i.e., three 768 different RTP sessions. In some scenarios, the same RTP stream may 769 be sent from the transmitting End Point, however it also supports 770 local adaptation taking place in one or more of the RTP streams, 771 rendering them non-identical. 773 +-A----------------------+ +-B-----------+ 774 |+---+ | | | 775 ||MIC| +-UDP1------| |-UDP1------+ | 776 |+---+ | +-RTP1----| |-RTP1----+ | | 777 | | +----+ | | +-Audio-| |-Audio-+ | | | 778 | +->|ENC1|--+-+-+--->AA1|------------->| | | | | 779 | | +----+ | | | |<-------------|BA1 | | | | 780 | | | | +-------| |-------+ | | | 781 | | | +---------| |---------+ | | 782 | | +-----------| |-----------+ | 783 | | ------------| |-------------| 784 | | | |-------------+ 785 | | | 786 | | | +-C-----------+ 787 | | | | | 788 | | +-UDP2------| |-UDP2------+ | 789 | | | +-RTP2----| |-RTP2----+ | | 790 | | +----+ | | +-Audio-| |-Audio-+ | | | 791 | +->|ENC2|--+-+-+--->AA2|------------->| | | | | 792 | +----+ | | | |<-------------|CA1 | | | | 793 | | | +-------| |-------+ | | | 794 | | +---------| |---------+ | | 795 | +-----------| |-----------+ | 796 +------------------------+ +-------------+ 798 Figure 10: An Multi-unicast Mesh with independent RTP session 800 Lets review the topology when independent RTP sessions are used, from 801 A's perspective in Figure 8 by considering both how the media is a 802 handled and the RTP sessions that are set-up in Figure 10. A's 803 microphone is captured and the digital audio can then be fed into two 804 different encoder instances, as each being associated with two 805 independent RTP sessions (RTP1 and RTP2). The SSRCs (AA1 and AA2) in 806 each RTP session are completely independent and the media bit-rate 807 produced by the encoders can also be tuned differently to address any 808 congestion control requirements differing for the paths A to B 809 compared to A to C. 811 From a topologies viewpoint, an important difference exists in the 812 behavior around RTCP. First, when a single RTP session spans all 813 three End Points A, B, and C, and their connecting RTP streams, a 814 common RTCP bandwidth is calculated and used for this single joint 815 session. In contrast, when there are multiple independent RTP 816 sessions, each RTP session has its local RTCP bandwidth allocation. 818 Further, when multiple sessions are used, End Points not directly 819 involved in a session do not have any awareness of the conditions in 820 those sessions. For example, in the case of the three End Point 821 configuration in Figure 8, End Point A has no awareness of the 822 conditions occurring in the session between End Points B and C 823 (whereas, if a single RTP session were used, it would have such 824 awareness). 826 Loop detection is also affected. With independent RTP sessions, the 827 SSRC/CSRC cannot be used to determine when an End Point receives its 828 own media stream, or a mixed media stream including its own media 829 stream (a condition known as a loop). The identification of loops 830 and, in most cases, their avoidance, has to be achieved by other 831 means, for example through signaling or the use of an RTP external 832 name space binding SSRC/CSRC among any communicating RTP sessions in 833 the mesh. 835 3.5. Point to Multipoint Using the RFC 3550 Translator 837 This section discusses some additional usages related to point to 838 multipoint of Translators compared to the point to point only cases 839 in Section 3.2.1. 841 3.5.1. Relay - Transport Translator 843 Shortcut name: Topo-PtM-Trn-Translator 845 This section discusses Transport Translator only usages to enable 846 multipoint sessions. 848 +-----+ 849 +---+ / \ +------------+ +---+ 850 | A |<---/ \ | |<---->| B | 851 +---+ / Multi- \ | | +---+ 852 + cast +->| Translator | 853 +---+ \ Network / | | +---+ 854 | C |<---\ / | |<---->| D | 855 +---+ \ / +------------+ +---+ 856 +-----+ 858 Figure 11: Point to Multipoint Using Multicast 860 Figure 11 depicts an example of a Transport Translator performing at 861 least IP address translation. It allows the (non-multicast-capable) 862 End Points B and D to take part in an any source multicast session 863 involving End Points A and C, by having the Translator forward their 864 unicast traffic to the multicast addresses in use, and vice versa. 865 It must also forward B's traffic to D, and vice versa, to provide 866 each of B and D with a complete view of the session. 868 +---+ +------------+ +---+ 869 | A |<---->| |<---->| B | 870 +---+ | | +---+ 871 | Translator | 872 +---+ | | +---+ 873 | C |<---->| |<---->| D | 874 +---+ +------------+ +---+ 876 Figure 12: RTP Translator (Relay) with Only Unicast Paths 878 Another Translator scenario is depicted in Figure 12. The Translator 879 in this case connects multiple End Points through unicast. This can 880 be implemented using a very simple transport Translator which, in 881 this document, is called a relay. The relay forwards all traffic it 882 receives, both RTP and RTCP, to all other End Points. In doing so, a 883 multicast network is emulated without relying on a multicast-capable 884 network infrastructure. 886 For RTCP feedback this results in a similar set of considerations to 887 those described in the ASM RTP topology. It also puts some 888 additional signalling requirements onto the session establishment; 889 for example, a common configuration of RTP payload types is required. 891 Transport translators and relays should always consider implementing 892 source address filtering, to prevent attackers to inject traffic 893 using the listening ports on the translator. The translator can, 894 however, go one step further, and especially if explicit SSRC 895 signalling is used, prevent End points to send SSRCs other than its 896 own (that are, for example, used by other participants in the 897 session). This can improve the security properties of the session, 898 despite the use of group keys that on cryptographic level allows 899 anyone to impersonate another in the same RTP session. 901 A Translator that doesn't change the RTP/RTCP packets content can be 902 operated without the requiring it to have access to the security 903 contexts used to protect the RTP/RTCP traffic between the 904 participants. 906 3.5.2. Media Translator 908 In the context of multipoint communications a Media Translator is not 909 providing new mechanisms to establish a multipoint session. It is 910 more of an enabler, or facilitator, that ensures a given End Point or 911 a defined sub-set of End Points can participate in the session. 913 If End Point B in Figure 11 were behind a limited network path, the 914 Translator may perform media transcoding to allow the traffic 915 received from the other End Points to reach B without overloading the 916 path. This transcoding can help the other End Points in the 917 multicast part of the session, by not requiring the quality 918 transmitted by A to be lowered to the bitrates that B is actually 919 capable of receiving (and vice versa). 921 3.6. Point to Multipoint Using the RFC 3550 Mixer Model 923 Shortcut name: Topo-Mixer 925 A Mixer is a middlebox that aggregates multiple RTP streams that are 926 part of a session by generating one or more new RTP streams and, in 927 most cases, by manipulating the media data. One common application 928 for a Mixer is to allow a participant to receive a session with a 929 reduced amount of resources. 931 +-----+ 932 +---+ / \ +-----------+ +---+ 933 | A |<---/ \ | |<---->| B | 934 +---+ / Multi- \ | | +---+ 935 + cast +->| Mixer | 936 +---+ \ Network / | | +---+ 937 | C |<---\ / | |<---->| D | 938 +---+ \ / +-----------+ +---+ 939 +-----+ 941 Figure 13: Point to Multipoint Using the RFC 3550 Mixer Model 943 A Mixer can be viewed as a device terminating the RTP streams 944 received from other End Points in the same RTP session. Using the 945 media data carried in the received RTP streams, a Mixer generates 946 derived RTP streams that are sent to the receiving End Points. 948 The content that the Mixer provides is the mixed aggregate of what 949 the Mixer receives over the PtP or PtM paths, which are part of the 950 same Communication Session. 952 The Mixer creates the Media Source and the source RTP stream just 953 like an End Point, as it mixes the content (often in the uncompressed 954 domain) and then encodes and packetizes it for transmission to a 955 receiving endpoint. The CSRC Count (CC) and CSRC fields in the RTP 956 header can be used to indicate the contributors to the newly 957 generated RTP stream. The SSRCs of the to-be-mixed streams on the 958 Mixer input appear as the CSRCs at the Mixer output. That output 959 stream uses a unique SSRC that identifies the Mixer's stream. The 960 CSRC should be forwarded between the different End Points to allow 961 for loop detection and identification of sources that are part of the 962 Communication Session. Note that Section 7.1 of RFC 3550 requires 963 the SSRC space to be shared between domains for these reasons. This 964 also implies that any SDES information normally needs to be forwarded 965 across the mixer. 967 The Mixer is responsible for generating RTCP packets in accordance 968 with its role. It is an RTP receiver and should therefore send RTCP 969 receiver reports for the RTP streams it receives and terminates. In 970 its role as an RTP sender, it should also generate RTCP sender 971 reports for those RTP streams it sends. As specified in Section 7.3 972 of RFC 3550, a Mixer must not forward RTCP unaltered between the two 973 domains. 975 The Mixer depicted in Figure 13 is involved in three domains that 976 need to be separated: the any source multicast network (including End 977 Points A and C), End Point B, and End Point D. Assuming all four End 978 Points in the conference are interested in receiving content from 979 each other End Point, the Mixer produces different mixed RTP streams 980 for B and D, as the one to B may contain content received from D, and 981 vice versa. However, the Mixer may only need one SSRC per media type 982 in each domain where it is the receiving entity and transmitter of 983 mixed content. 985 In the multicast domain, a Mixer still needs to provide a mixed view 986 of the other domains. This makes the Mixer simpler to implement and 987 avoids any issues with advanced RTCP handling or loop detection, 988 which would be problematic if the Mixer were providing non-symmetric 989 behavior. Please see Section 3.11 for more discussion on this topic. 990 The mixing operation, however, in each domain could potentially be 991 different. 993 A Mixer is responsible for receiving RTCP feedback messages and 994 handling them appropriately. The definition of "appropriate" depends 995 on the message itself and the context. In some cases, the reception 996 of a codec-control message by the Mixer may result in the generation 997 and transmission of RTCP feedback messages by the Mixer to the End 998 Points in the other domain(s). In other cases, a message is handled 999 by the Mixer locally and therefore not forwarded to any other domain. 1001 When replacing the multicast network in Figure 13 (to the left of the 1002 Mixer) with individual unicast paths as depicted in Figure 14, the 1003 Mixer model is very similar to the one discussed in Section 3.9 1004 below. Please see the discussion in Section 3.9 about the 1005 differences between these two models. 1007 +---+ +------------+ +---+ 1008 | A |<---->| |<---->| B | 1009 +---+ | | +---+ 1010 | Mixer | 1011 +---+ | | +---+ 1012 | C |<---->| |<---->| D | 1013 +---+ +------------+ +---+ 1015 Figure 14: RTP Mixer with Only Unicast Paths 1017 We now discuss in more detail the different mixing operations that a 1018 mixer can perform and how they can affect RTP and RTCP behavior. 1020 3.6.1. Media Mixing Mixer 1022 The media mixing mixer is likely the one that most think of when they 1023 hear the term "mixer". Its basic mode of operation is that it 1024 receives RTP streams from several End Points and selects the 1025 stream(s) to be included in a media-domain mix. The selection can be 1026 through static configuration or by dynamic, content dependent means 1027 such as voice activation. The mixer then creates a single outgoing 1028 RTP stream from this mix. 1030 The most commonly deployed media mixer is probably the audio mixer, 1031 used in voice conferencing, where the output consists of a mixture of 1032 all the input audio signals; this needs minimal signalling to be 1033 successfully set up. From a signal processing viewpoint, audio 1034 mixing is relatively straightforward and commonly possible for a 1035 reasonable number of End Points. Assume, for example, that one wants 1036 to mix N streams from N different End Points. The mixer needs to 1037 decode those N streams, typically into the sample domain, and then 1038 produce N or N+1 mixes. Different mixes are needed so that each 1039 contributing source gets a mix of all other sources except its own, 1040 as this would result in an echo. When N is lower than the number of 1041 all End points, one may produce a mix of all N streams for the group 1042 that are currently not included in the mix, thus N+1 mixes. These 1043 audio streams are then encoded again, RTP packetized and sent out. 1044 In many cases, audio level normalization, noise suppression, and 1045 similar signal processing steps are also required or desirable before 1046 the actual mixing process commences. 1048 In video, the term "mixing" has a different interpretation than 1049 audio. It is commonly used to refer to the process of spatially 1050 combining contributed video streams, which is also known as "tiling". 1051 The reconstructed, appropriately scaled down videos can be spatially 1052 arranged in a set of tiles, each tile containing the video from an 1053 End Point (typically showing a human participant). Tiles can be of 1054 different sizes, so that, for example, a particularly important 1055 participant, or the loudest speaker, is being shown on in larger tile 1056 than other participants. A self-view picture can be included in the 1057 tiling, which can either be locally produced or be a feedback from a 1058 mixer-received and reconstructed video image. Such remote loopback 1059 allows for confidence monitoring, i.e., it enables the participant to 1060 see himself/herself in the same quality as other participants see 1061 him/her. The tiling normally operates on reconstructed video in the 1062 sample domain. The tiled image is encoded, packetized, and sent by 1063 the mixer to the receiving End Points. It is possible that a 1064 middlebox with media mixing duties contains only a single mixer of 1065 the aforementioned type, in which case all participants necessarily 1066 see the same tiled video, even if it is being sent over different RTP 1067 streams. More common, however, are mixing arrangement where an 1068 individual mixer is available for each outgoing port of the 1069 middlebox, allowing individual compositions for each receiving End 1070 Point (a feature commonly referred to as personalized layout). 1072 One problem with media mixing is that it consumes both large amounts 1073 of media processing resources (for the decoding and mixing process in 1074 the uncompressed domain) and encoding resources (for the encoding of 1075 the mixed signal). Another problem is the quality degradation 1076 created by decoding and re-encoding the media, which is the result of 1077 the lossy nature of most commonly used media codecs. A third problem 1078 is the latency introduced by the media mixing, which can be 1079 substantial and annoyingly noticeable in case of video, or in case of 1080 audio if that mixed audio is lip-sychronized with high latency video. 1081 The advantage of media mixing is that it is straightforward for the 1082 End Points to handle the single media stream (which includes the 1083 mixed aggregate of many sources), as they don't need to handle 1084 multiple decodings, local mixing and composition. In fact, mixers 1085 were introduced in pre-RTP times so that legacy, single stream 1086 receiving endpoints (that, in some protocol environments, actually 1087 didn't need to be aware of the multipoint nature of teh conference) 1088 could successfully participate in what a user would recognize as a 1089 multiparty video conference. 1091 +-A---------+ +-MIXER----------------------+ 1092 | +-RTP1----| |-RTP1------+ +-----+ | 1093 | | +-Audio-| |-Audio---+ | +---+ | | | 1094 | | | AA1|--------->|---------+-+-|DEC|->| | | 1095 | | | |<---------|MA1 <----+ | +---+ | | | 1096 | | | | |(BA1+CA1)|\| +---+ | | | 1097 | | +-------| |---------+ +-|ENC|<-| B+C | | 1098 | +---------| |-----------+ +---+ | | | 1099 +-----------+ | | | | 1100 | | M | | 1101 +-B---------+ | | E | | 1102 | +-RTP2----| |-RTP2------+ | D | | 1103 | | +-Audio-| |-Audio---+ | +---+ | I | | 1104 | | | BA1|--------->|---------+-+-|DEC|->| A | | 1105 | | | |<---------|MA2 <----+ | +---+ | | | 1106 | | +-------| |(BA1+CA1)|\| +---+ | | | 1107 | +---------| |---------+ +-|ENC|<-| A+C | | 1108 +-----------+ |-----------+ +---+ | | | 1109 | | M | | 1110 +-C---------+ | | I | | 1111 | +-RTP3----| |-RTP3------+ | X | | 1112 | | +-Audio-| |-Audio---+ | +---+ | E | | 1113 | | | CA1|--------->|---------+-+-|DEC|->| R | | 1114 | | | |<---------|MA3 <----+ | +---+ | | | 1115 | | +-------| |(BA1+CA1)|\| +---+ | | | 1116 | +---------| |---------+ +-|ENC|<-| A+B | | 1117 +-----------+ |-----------+ +---+ +-----+ | 1118 +----------------------------+ 1120 Figure 15: Session and SSRC details for Media Mixer 1122 From an RTP perspective media mixing can be a very simple process, as 1123 can be seen in Figure 15. The mixer presents one SSRC towards the 1124 receiving End Point, e.g., MA1 to Peer A, where the associated stream 1125 is the media mix of the other End Points. As each peer, in this 1126 example, receives a different version of a mix from the mixer, there 1127 is no actual relation between the different RTP sessions in terms of 1128 actual media or transport level information. There are, however, 1129 common relationships between RTP1-RTP3, namely SSRC space and 1130 identity information. When A receives the MA1 stream which is a 1131 combination of BA1 and CA1 streams, the mixer may include CSRC 1132 information in the MA1 stream to identify the contributing source BA1 1133 and CA1, allowing the receiver to identify the contributing sources 1134 even if this were not possible through the media itself or through 1135 other signaling means. 1137 The CSRC has, in turn, utility in RTP extensions, like the Mixer to 1138 Client audio levels RTP header extension [RFC6465]. If the SSRCs 1139 from the End Point to mixer paths are used as CSRCs in another RTP 1140 session, then RTP1, RTP2 and RTP3 become one joint session as they 1141 have a common SSRC space. At this stage, the mixer also needs to 1142 consider which RTCP information it needs to expose in the different 1143 paths. In the above scenario, a mixer would normally expose nothing 1144 more than the Source Description (SDES) information and RTCP BYE for 1145 a CSRC leaving the session. The main goal would be to enable the 1146 correct binding against the application logic and other information 1147 sources. This also enables loop detection in the RTP session. 1149 3.6.2. Media Switching 1151 Media switching mixers are used in limited functionality scenarios 1152 where no, or only very limited, concurrent presentation of multiple 1153 sources is required by the application, to more complex multi-stream 1154 usages with receiver mixing or tiling, including combined with 1155 simulcast and/or scalability between source and mixer. An RTP Mixer 1156 based on media switching avoids the media decoding and encoding 1157 operations in the mixer, as it conceptually forwards the encoded 1158 media stream as it was being sent to the mixer. It does not avoid, 1159 however, the decryption and re-encryption cycle as it rewrites RTP 1160 headers. Forwarding media (in contrast to reconstructing-mixing- 1161 encoding media) reduces the amount of computational resources needed 1162 in the mixer and increases the media quality (both in terms of 1163 fidelity and reduced latency). 1165 A media switching mixer maintains a pool of SSRCs representing 1166 conceptual or functional RTP streams that the mixer can produce. 1167 These RTP streams are created by selecting media from one of the RTP 1168 streams received by the mixer and forwarded to the peer using the 1169 mixer's own SSRCs. The mixer can switch between available sources if 1170 that is required by the concept for the source, like the currently 1171 active speaker. Note that the mixer, in most cases, still needs to 1172 perform a certain amount of media processing, as many media formats 1173 do not allow to "tune into" the stream at arbitrary points in their 1174 bitstream. 1176 To achieve a coherent RTP stream from the mixer's SSRC, the mixer 1177 needs to rewrite the incoming RTP packet's header. First the SSRC 1178 field must be set to the value of the Mixer's SSRC. Second, the 1179 sequence number must be the next in the sequence of outgoing packets 1180 it sent. Third, the RTP timestamp value needs to be adjusted using 1181 an offset that changes each time one switches media source. Finally, 1182 depending on the negotiation of the RTP payload type, the value 1183 representing this particular RTP payload configuration may have to be 1184 changed if the different End Point-to-mixer paths have not arrived on 1185 the same numbering for a given configuration. This also requires 1186 that the different End Points support a common set of codecs, 1187 otherwise media transcoding for codec compatibility would still be 1188 required. 1190 We now consider the operation of a media switching mixer that 1191 supports a video conference with six participating End Points (A-F) 1192 where the two most recent speakers in the conference are shown to 1193 each receiving End Point. The mixer has thus two SSRCs sending video 1194 to each peer, and each peer is capable of locally handling two video 1195 streams simultaneously. 1197 +-A---------+ +-MIXER----------------------+ 1198 | +-RTP1----| |-RTP1------+ +-----+ | 1199 | | +-Video-| |-Video---+ | | | | 1200 | | | AV1|------------>|---------+-+------->| S | | 1201 | | | |<------------|MV1 <----+-+-BV1----| W | | 1202 | | | |<------------|MV2 <----+-+-EV1----| I | | 1203 | | +-------| |---------+ | | T | | 1204 | +---------| |-----------+ | C | | 1205 +-----------+ | | H | | 1206 | | | | 1207 +-B---------+ | | M | | 1208 | +-RTP2----| |-RTP2------+ | A | | 1209 | | +-Video-| |-Video---+ | | T | | 1210 | | | BV1|------------>|---------+-+------->| R | | 1211 | | | |<------------|MV3 <----+-+-AV1----| I | | 1212 | | | |<------------|MV4 <----+-+-EV1----| X | | 1213 | | +-------| |---------+ | | | | 1214 | +---------| |-----------+ | | | 1215 +-----------+ | | | | 1216 : : : : 1217 : : : : 1218 +-F---------+ | | | | 1219 | +-RTP6----| |-RTP6------+ | | | 1220 | | +-Video-| |-Video---+ | | | | 1221 | | | CV1|------------>|---------+-+------->| | | 1222 | | | |<------------|MV11 <---+-+-AV1----| | | 1223 | | | |<------------|MV12 <---+-+-EV1----| | | 1224 | | +-------| |---------+ | | | | 1225 | +---------| |-----------+ +-----+ | 1226 +-----------+ +----------------------------+ 1228 Figure 16: Media Switching RTP Mixer 1230 The Media Switching RTP mixer can, similarly to the Media Mixing 1231 Mixer, reduce the bit-rate required for media transmission towards 1232 the different peers by selecting and forwarding only a sub-set of RTP 1233 streams it receives from the sending End Points. In cases the mixer 1234 receives simulcast transmissions or a scalable encoding of the media 1235 source, the mixer has more degrees of freedom to select streams or 1236 sub-sets of stream to forward to a receiving End Point, both based on 1237 transport or End Point restrictions as well as application logic. 1239 To ensure that a media receiver in an End Point can correctly decode 1240 the media in the RTP stream after a switch, a codec that uses 1241 temporal prediction needs to start its decoding from independent 1242 refresh points, or points in the bitstream offering similar 1243 functionality (like "dirty refresh points"). For some codecs, for 1244 example frame based speech and audio codecs, this is easily achieved 1245 by starting the decoding at RTP packet boundaries, as each packet 1246 boundary provides a refresh point (assuming proper packetization on 1247 the encoder side). For other codecs, particularly in video, refresh 1248 points are less common in the bitstream or may not be present at all 1249 without an explicit request to the respective encoder. The Full 1250 Intra Request [RFC5104] RTCP codec control message has been defined 1251 for this purpose. 1253 In this type of mixer one could consider to fully terminate the RTP 1254 sessions between the different End Point and mixer paths. The same 1255 arguments and considerations as discussed in Section 3.9 need to be 1256 taken into consideration and apply here. 1258 3.7. Selective Forwarding Middlebox 1260 Another method for handling media in the RTP mixer is to "project", 1261 or make available, all potential RTP sources (SSRCs) into a per-End 1262 Point, independent RTP session. The middlebox can select which of 1263 the potential sources that are currently actively transmitting media 1264 will be sent to each of the End Points. This is similar to the media 1265 switching Mixer but has some important differences in RTP details. 1267 +-A---------+ +-Middlebox-----------------+ 1268 | +-RTP1----| |-RTP1------+ +-----+ | 1269 | | +-Video-| |-Video---+ | | | | 1270 | | | AV1|------------>|---------+-+------>| | | 1271 | | | |<------------|BV1 <----+-+-------| S | | 1272 | | | |<------------|CV1 <----+-+-------| W | | 1273 | | | |<------------|DV1 <----+-+-------| I | | 1274 | | | |<------------|EV1 <----+-+-------| T | | 1275 | | | |<------------|FV1 <----+-+-------| C | | 1276 | | +-------| |---------+ | | H | | 1277 | +---------| |-----------+ | | | 1278 +-----------+ | | M | | 1279 | | A | | 1280 +-B---------+ | | T | | 1281 | +-RTP2----| |-RTP2------+ | R | | 1282 | | +-Video-| |-Video---+ | | I | | 1283 | | | BV1|------------>|---------+-+------>| X | | 1284 | | | |<------------|AV1 <----+-+-------| | | 1285 | | | |<------------|CV1 <----+-+-------| | | 1286 | | | | : : : |: : : : : : : : :| | | 1287 | | | |<------------|FV1 <----+-+-------| | | 1288 | | +-------| |---------+ | | | | 1289 | +---------| |-----------+ | | | 1290 +-----------+ | | | | 1291 : : : : 1292 : : : : 1293 +-F---------+ | | | | 1294 | +-RTP6----| |-RTP6------+ | | | 1295 | | +-Video-| |-Video---+ | | | | 1296 | | | FV1|------------>|---------+-+------>| | | 1297 | | | |<------------|AV1 <----+-+-------| | | 1298 | | | | : : : |: : : : : : : : :| | | 1299 | | | |<------------|EV1 <----+-+-------| | | 1300 | | +-------| |---------+ | | | | 1301 | +---------| |-----------+ +-----+ | 1302 +-----------+ +---------------------------+ 1304 Figure 17: Selective Forwarding Middlebox 1306 In the six End Point conference depicted above in (Figure 17) one can 1307 see that End Point A is aware of five incoming SSRCs, BV1-FV1. If 1308 this middlebox intends to have a similar behavior as in Section 3.6.2 1309 where the mixer provides the End Points with the two latest speaking 1310 End Points, then only two out of these five SSRCs need concurrently 1311 transmit media to A. As the middlebox selects the source in the 1312 different RTP sessions that transmit media to the End points, each 1313 RTP stream requires rewriting of certain RTP header fields when being 1314 projected from one session into another. In particular, the sequence 1315 number needs to be consecutively incremented based on the packet 1316 actually being transmitted in each RTP session. Therefore, the RTP 1317 sequence number offset will change each time a source is turned on in 1318 a RTP session. The timestamp (possibly offset) stays the same. 1320 As the RTP sessions are independent, the SSRC numbers used can also 1321 be handled independently, thereby bypassing the requirement for SSRC 1322 collision detection and avoidance. On the other hand, tools such as 1323 remapping tables between the RTP sessions are required. For example, 1324 the RTP stream that is being sent by End Point B to the middlebox 1325 (BV1) may use an SSRC value of 12345678. When that RTP stream is 1326 sent to End Point F by the middlebox, it can use any SSRC value, e.g. 1327 87654321. As a result, each End Point may have a different view of 1328 the application usage of a particular SSRC. Any RTP level identity 1329 information, such as SDES items also needs to update the SSRC 1330 referenced, if the included SDES items are intended to be global. 1331 Thus the application must not use SSRC as references to RTP streams 1332 when communicating with other peers directly. This also affects loop 1333 detection which will fail to work, as there is no common namespace 1334 and identities across the different legs in the communication session 1335 on RTP level. Instead this responsibility falls onto higher layers. 1337 The middlebox is also responsible to receive any RTCP codec control 1338 requests coming from an End Point, and decide if it can act on the 1339 request locally or needs to translate the request into the RTP 1340 session that contains the media source. Both End Points and the 1341 middlebox need to implement conference related codec control 1342 functionalities to provide a good experience. Commonly used are Full 1343 Intra Request to request from the media source to provide switching 1344 points between the sources, and Temporary Maximum Media Bit-rate 1345 Request (TMMBR) to enable the middlebox to aggregate congestion 1346 control responses towards the media source so to enable it to adjust 1347 its bit-rate (obviously only in case the limitation is not in the 1348 source to middlebox link). 1350 The selective forwarding middlebox has been introduced in recently 1351 developed videoconferencing systems in conjunction with, and to 1352 capitalize on, scalable video coding as well as simulcasting. An 1353 example of scalable video coding is Annex G of H.264, but other 1354 codecs, including H.264 AVC and VP8 also exhibit scalability, albeit 1355 only in the temporal dimension. In both scalable coding and 1356 simulcast cases the video signal is represented by a set of two or 1357 more bitstreams, providing a corresponding number of distinct 1358 fidelity points. The middlebox selects which parts of a scalable 1359 bitstream (or which bitstream, in the case of simulcasting) to 1360 forward to each of the receiving End Points. The decision may be 1361 driven by a number of factors, such as available bit rate, desired 1362 layout, etc. Contrary to transcoding MCUs, these "Selective 1363 Forwarding Units" (SFUs) have extremely low delay, and provide 1364 features that are typically associated with high-end systems 1365 (personalized layout, error localization) without any signal 1366 processing at the middlebox. They are also capable of scaling to a 1367 large number of concurrent users, and--due to their very low delay-- 1368 can also be cascaded. 1370 This version of the middlebox also puts different requirements on the 1371 End Point when it comes to decoder instances and handling of the RTP 1372 streams providing media. As each projected SSRC can, at any time, 1373 provide media, the End Point either needs to be able to handle as 1374 many decoder instances as the middlebox received, or have efficient 1375 switching of decoder contexts in a more limited set of actual decoder 1376 instances to cope with the switches. The application also gets more 1377 responsibility to update how the media provided is to be presented to 1378 the user. 1380 Note that this topology could potentially be seen as a media 1381 translator which include an on/off logic as part of its media 1382 translation. The main difference would be a common global SSRC space 1383 in the case of the Media Translator and the mapped one used in the 1384 above. It also has mixer aspects, as the streams it provides are not 1385 basically translated version, but instead they have conceptual 1386 property assigned to them. Thus this topology appears to be some 1387 hybrid between the translator and mixer model. 1389 The differences between selective forwarding middlebox and a 1390 switching mixer (Section 3.6.2) are minor, and they share most 1391 properties. The above requirement on having a large number of 1392 decoding instances or requiring efficient switching of decoder 1393 contexts, are one point of difference. The other is how the 1394 identification is performed, where the Mixer uses CSRC to provide 1395 information on what is included in a particular RTP stream that 1396 represent a particular concept. Selective forwarding gets the source 1397 information through the SSRC, and instead have to use other mechanism 1398 to make clear the streams current purpose. 1400 3.8. Point to Multipoint Using Video Switching MCUs 1402 Shortcut name: Topo-Video-switch-MCU 1403 +---+ +------------+ +---+ 1404 | A |------| Multipoint |------| B | 1405 +---+ | Control | +---+ 1406 | Unit | 1407 +---+ | (MCU) | +---+ 1408 | C |------| |------| D | 1409 +---+ +------------+ +---+ 1411 Figure 18: Point to Multipoint Using a Video Switching MCU 1413 This PtM topology was popular in early implementations of multipoint 1414 videoconferencing systems due to its simplicity, and the 1415 corresponding middlebox design has been known as a "video switching 1416 MCU". The more complex RTCP-terminating MCUs, discussed in the next 1417 section, became the norm, however, when technology allowed 1418 implementations at acceptable costs. 1420 A video switching MCU forwards to a participant a single media 1421 stream, selected from the available streams. The criteria for 1422 selection are often based on voice activity in the audio-visual 1423 conference, but other conference management mechanisms (like 1424 presentation mode or explicit floor control) are known to exist as 1425 well. 1427 The video switching MCU may also perform media translation to modify 1428 the content in bit-rate, encoding, or resolution. However, it still 1429 may indicate the original sender of the content through the SSRC. In 1430 this case, the values of the CC and CSRC fields are retained. 1432 If not terminating RTP, the RTCP Sender Reports are forwarded for the 1433 currently selected sender. All RTCP Receiver Reports are freely 1434 forwarded between the End points. In addition, the MCU may also 1435 originate RTCP control traffic in order to control the session and/or 1436 report on status from its viewpoint. 1438 The video switching MCU has most of the attributes of a Translator. 1439 However, its stream selection is a mixing behavior. This behavior 1440 has some RTP and RTCP issues associated with it. The suppression of 1441 all but one RTP stream results in most participants seeing only a 1442 subset of the sent RTP streams at any given time, often a single RTP 1443 stream per conference. Therefore, RTCP Receiver Reports only report 1444 on these RTP streams. Consequently, the End Points emitting RTP 1445 streams that are not currently forwarded receive a view of the 1446 session that indicates their RTP streams disappear somewhere en 1447 route. This makes the use of RTCP for congestion control, or any 1448 type of quality reporting, very problematic. 1450 To avoid the aforementioned issues, the MCU needs to implement two 1451 features. First, it needs to act as a Mixer (see Section 3.6) and 1452 forward the selected RTP stream under its own SSRC and with the 1453 appropriate CSRC values. Second, the MCU needs to modify the RTCP 1454 RRs it forwards between the domains. As a result, it is recommended 1455 that one implement a centralized video switching conference using a 1456 Mixer according to RFC 3550, instead of the shortcut implementation 1457 described here. 1459 3.9. Point to Multipoint Using RTCP-Terminating MCU 1461 Shortcut name: Topo-RTCP-terminating-MCU 1463 +---+ +------------+ +---+ 1464 | A |<---->| Multipoint |<---->| B | 1465 +---+ | Control | +---+ 1466 | Unit | 1467 +---+ | (MCU) | +---+ 1468 | C |<---->| |<---->| D | 1469 +---+ +------------+ +---+ 1471 Figure 19: Point to Multipoint Using Content Modifying MCUs 1473 In this PtM scenario, each End Point runs an RTP point-to-point 1474 session between itself and the MCU. This is a very commonly deployed 1475 topology in multipoint video conferencing. The content that the MCU 1476 provides to each participant is either: 1478 a. a selection of the content received from the other End Points, or 1480 b. the mixed aggregate of what the MCU receives from the other PtP 1481 paths, which are part of the same Communication Session. 1483 In case (a), the MCU may modify the content in terms of bit-rate, 1484 encoding format, or resolution. No explicit RTP mechanism is used to 1485 establish the relationship between the original RTP stream of the 1486 media being sent RTP stream the MCU sends. In other words, the 1487 outgoing RTP streams typically use a different SSRC, and may well use 1488 a different payload type (PT), even if this different PT happens to 1489 be mapped to the same media type. This is a result of the 1490 individually negotiated RTP session for each End Point. 1492 In case (b), the MCU is the Media Source and generates the Source RTP 1493 Stream as it mixes the received content and then encodes and 1494 packetizes it for transmission to an End Point. According to RTP 1495 [RFC3550], the SSRC of the contributors are to be signalled using the 1496 CSRC/CC mechanism. In practice, today, most deployed MCUs do not 1497 implement this feature. Instead, the identification of the End 1498 Points whose content is included in the Mixer's output is not 1499 indicated through any explicit RTP mechanism. That is, most deployed 1500 MCUs set the CSRC Count (CC) field in the RTP header to zero, thereby 1501 indicating no available CSRC information, even if they could identify 1502 the original sending End Points as suggested in RTP. 1504 The main feature that sets this topology apart from what RFC 3550 1505 describes is the breaking of the common RTP session across the 1506 centralized device, such as the MCU. This results in the loss of 1507 explicit RTP-level indication of all participants. If one were using 1508 the mechanisms available in RTP and RTCP to signal this explicitly, 1509 the topology would follow the approach of an RTP Mixer. The lack of 1510 explicit indication has at least the following potential problems: 1512 1. Loop detection cannot be performed on the RTP level. When 1513 carelessly connecting two misconfigured MCUs, a loop could be 1514 generated. 1516 2. There is no information about active media senders available in 1517 the RTP packet. As this information is missing, receivers cannot 1518 use it. It also deprives the client of information related to 1519 currently active senders in a machine-usable way, thus preventing 1520 clients from indicating currently active speakers in user 1521 interfaces, etc. 1523 Note that many/most deployed MCUs (and video conferencing endpoints) 1524 rely on signalling layer mechanisms for the identification of the 1525 contributing sources, for example, a SIP conferencing package 1526 [RFC4575]. This alleviates, to some extent, the aforementioned 1527 issues resulting from ignoring RTP's CSRC mechanism. 1529 3.10. Split Component Terminal 1531 Shortcut name: Topo-Split-Terminal 1533 In some applications, for example in some telepresence systems, 1534 terminals may be not integrated into a single functional unit, but 1535 composed of more than one subunits. For example, a telepresence room 1536 terminal employing multiple cameras and monitors may consist of 1537 multiple video conferencing subunits, each capable of handling a 1538 single camera and monitor. Another example would be a video 1539 conferencing terminal in which audio is handled by one subunit, and 1540 video by another. Each of these subunits uses its own network 1541 address. The various (media processing) subunits need (logically and 1542 physically) to be interconnected by control functionality, but their 1543 media plane functionality may be split. This type of terminals is 1544 referred to as split component terminals. 1546 An example for such a split component terminal is depicted in 1547 Figure 20. Within split component terminal A, at least audio and 1548 video subunits are addressed by their own network addresses. In some 1549 of these systems, the control stack subunit may also have its own 1550 network address. 1552 From an RTP viewpoint, each of the subunits terminate RTP, and act as 1553 an End Point in the sense that each subunit includes its own, 1554 independent RTP stack. However, as the subunits are semantically 1555 part of the same terminal, it is appropriate that this semantic 1556 relationship is expressed in RTCP protocol elements, namely in the 1557 CNAME. 1559 +---------------------+ 1560 | Endpoint A | 1561 | Local Area Network | 1562 | +------------+ | 1563 | +->| Audio |<+-RTP---\ 1564 | | +------------+ | \ +------+ 1565 | | +------------+ | +-->| | 1566 | +->| Video |<+-RTP-------->| B | 1567 | | +------------+ | +-->| | 1568 | | +------------+ | / +------+ 1569 | +->| Control |<+-SIP---/ 1570 | +------------+ | 1571 +---------------------+ 1573 Figure 20: Split Component Terminal 1575 In a compliant RTP implementation, it is not feasible for more than 1576 one subunit to be part of a given RTP session. When attempting to do 1577 so, a third party monitor would report the two subunits as two 1578 separate End Points with a CNAME collision. As a result, a fully RTP 1579 conformant split component terminal is one where the subunits use 1580 separate RTP sessions to send and/or receive RTP streams intended for 1581 them. 1583 In addition to the use of a common CNAME and the use of independent 1584 RTP sessions for the RTP streams generated or consumed by the various 1585 subunits, it is sensible that the subunits share a common clock, to 1586 ensure that synchronization and clock drift handling works, despite 1587 the fact that the components are separated. RTCP handling works 1588 correctly as long as each subunit participates in its own RTP 1589 session. 1591 3.11. Non-Symmetric Mixer/Translators 1593 Shortcut name: Topo-Asymmetric 1595 It is theoretically possible to construct an MCU that is a Mixer in 1596 one direction and a Translator in another. The main reason to 1597 consider this would be to allow topologies similar to Figure 13, 1598 where the Mixer does not need to mix in the direction from B or D 1599 towards the multicast domains with A and C. Instead, the RTP streams 1600 from B and D are forwarded without changes. Avoiding this mixing 1601 would save media processing resources that perform the mixing in 1602 cases where it isn't needed. However, there would still be a need to 1603 mix B's media towards D. Only in the direction B -> multicast domain 1604 or D -> multicast domain would it be possible to work as a 1605 Translator. In all other directions, it would function as a Mixer. 1607 The Mixer/Translator would still need to process and change the RTCP 1608 before forwarding it in the directions of B or D to the multicast 1609 domain. One issue is that A and C do not know about the mixed-media 1610 stream the Mixer sends to either B or D. Therefore, any reports 1611 related to these streams must be removed. Also, receiver reports 1612 related to A and C's RTP streams would be missing. To avoid A and C 1613 thinking that B and D aren't receiving A and C at all, the Mixer 1614 needs to insert locally generated reports reflecting the situation 1615 for the streams from A and C into B and D's Sender Reports. In the 1616 opposite direction, the Receiver Reports from A and C about B's and 1617 D's stream also need to be aggregated into the Mixer's Receiver 1618 Reports sent to B and D. Since B and D only have the Mixer as source 1619 for the stream, all RTCP from A and C must be suppressed by the 1620 Mixer. 1622 This topology is so problematic and it is so easy to get the RTCP 1623 processing wrong, that it is not recommended for implementation. 1625 3.12. Combining Topologies 1627 Topologies can be combined and linked to each other using Mixers or 1628 Translators. However, care must be taken in handling the SSRC/CSRC 1629 space. A Mixer does not forward RTCP from sources in other domains, 1630 but instead generates its own RTCP packets for each domain it mixes 1631 into, including the necessary Source Description (SDES) information 1632 for both the CSRCs and the SSRCs. Thus, in a mixed domain, the only 1633 SSRCs seen will be the ones present in the domain, while there can be 1634 CSRCs from all the domains connected together with a combination of 1635 Mixers and Translators. The combined SSRC and CSRC space is common 1636 over any Translator or Mixer. It is important to facilitate loop 1637 detection, something that is likely to be even more important in 1638 combined topologies due to the mixed behavior between the domains. 1640 Any hybrid, like the Topo-Video-switch-MCU or Topo-Asymmetric, 1641 requires considerable thought on how RTCP is dealt with. 1643 4. Comparing Topologies 1645 The topologies discussed in Section 3 have different properties. 1646 This section first describes these properties and then analyzes how 1647 these properties are supported by the different topologies. Note 1648 that, even if a certain property is supported within a particular 1649 topology concept, the necessary functionality may be optional to 1650 implement. 1652 4.1. Topology Properties 1654 4.1.1. All to All Media Transmission 1656 To recapitulate, multicast, and in particular Any Source Multicast 1657 (ASM), provides the functionality that everyone may send to, or 1658 receive from, everyone else within the session. Source-specific 1659 Multicast (SSM) can provide a similar functionality by having anyone 1660 intending to participate as sender to send its media to the SSM 1661 distribution source. The SSM distribution source forwards the media 1662 to all receivers subscribed to the multicast group. Mesh, MCUs, 1663 Mixers, SFMs and Translators may all provide that functionality at 1664 least on some basic level. However, there are some differences in 1665 which type of reachability they provide. 1667 Closest to true IP-multicast-based, all-to-all transmission comes 1668 perhaps the transport Translator function called "relay" in in 1669 Section 3.5, as well as the Mesh with joint RTP sessions. Media 1670 Translators, Mesh with independent RTP Sessions, Mixers, SFUs and the 1671 MCU variants do not provide a fully meshed forwarding on the 1672 transport level; instead, they only allow limited forwarding of 1673 content from the other session participants. 1675 The "all to all media transmission" requires that any media 1676 transmitting End Point considers the path to the least capable 1677 receiving End Point. Otherwise, the media transmissions may overload 1678 that path. Therefore, a sending End Point needs to monitor the path 1679 from itself to any of the receiving End Points, to detect the 1680 currently least capable receiver, and adapt its sending rate 1681 accordingly. As multiple End Points may send simultaneously, the 1682 available resources may vary. RTCP's Receiver Reports help 1683 performing this monitoring, at least on a medium time scale. 1685 The resource consumption for performing all to all transmission 1686 varies depending with the topology. Both ASM and SSM have the 1687 benefit that only one copy of each packet traverses a particular 1688 link. Using a relay causes the transmission of one copy of a packet 1689 per End Point-to-relay path and packet transmitted. However, in most 1690 cases the links carrying the multiple copies will be the ones close 1691 to the relay (which can be assumed to be part of the network 1692 infrastructure with good connectivity to the backbone), rather than 1693 the End Points (which may be behind slower access links). The Mesh 1694 causes N-1 streams of transmitted packets to traverse the first hop 1695 link from the End Point, in an N End Point mesh. How long the 1696 different paths are common, is highly situation dependent. 1698 The transmission of RTCP by design adapts to any changes in the 1699 number of participants due to the transmission algorithm, defined in 1700 the RTP specification [RFC3550], and the extensions in AVPF [RFC4585] 1701 (when applicable). That way, the resources utilized for RTCP stay 1702 within the bounds configured for the session. 1704 4.1.2. Transport or Media Interoperability 1706 All Translators, Mixers, and RTCP-terminating MCU, and Mesh with 1707 individual RTP sessions, allow changing the media encoding or the 1708 transport to other properties of the other domain, thereby providing 1709 extended interoperability in cases where the End Points lack a common 1710 set of media codecs and/or transport protocols. Selective Forwarding 1711 Middleboxes can adopt the transport, and (at least) selectively 1712 forward the encoded streams that match a receiving End Point's 1713 capability. It requires an additional translator to change the media 1714 encoding if the encoded streams do not match the receiving End 1715 Point's capabilities. 1717 4.1.3. Per Domain Bit-Rate Adaptation 1719 End Points are often connected to each other with a heterogeneous set 1720 of paths. This makes congestion control in a Point to Multipoint set 1721 problematic. For the ASM, SSM, Mesh with common RTP session, and 1722 Transport Relay scenario, each individual sending End Point has to 1723 adapt to the receiving End Point behind the least capable path, 1724 yielding suboptimal quality for the End Points behind the more 1725 capable paths. This is no longer an issue when Media Translators, 1726 Mixers, SFM or MCUs are involved, as each End Point only needs to 1727 adapt to the slowest path within its own domain. The Translator, 1728 Mixer, SFM, or MCU topologies all require their respective outgoing 1729 RTP streams to adjust the bit-rate, packet-rate, etc., to adapt to 1730 the least capable path in each of the other domains. That way one 1731 can avoid lowering the quality to the least-capable End Point in all 1732 the domains at the cost (complexity, delay, equipment) of the Mixer, 1733 SFM or Translator, and potentially media sender (multicast/layered 1734 encoding and sending the different representations). 1736 4.1.4. Aggregation of Media 1738 In the all-to-all media property mentioned above and provided by ASM, 1739 SSM, Mesh with common RTP session, and relay, all simultaneous media 1740 transmissions share the available bit-rate. For End Points with 1741 limited reception capabilities, this may result in a situation where 1742 even a minimal acceptable media quality cannot be accomplished, 1743 because multiple RTP streams need to share the same resources. One 1744 solution to this problem is to provide for a Mixer, or MCU to 1745 aggregate the multiple RTP streams into a single one, where the 1746 single RTP stream takes up less resources in terms of bit-rate. This 1747 aggregation can be performed according to different methods. Mixing 1748 or selection are two common methods. Selection is almost always 1749 possible and easy to implement. Mixing requires resources in the 1750 mixer, and may be relatively easy and not impairing the quality too 1751 badly (audio) or quite difficult (video tiling, which is not only 1752 computationally complex but also reduces the pixel count per stream, 1753 with corresponding loss in perceptual quality). 1755 4.1.5. View of All Session Participants 1757 The RTP protocol includes functionality to identify the session 1758 participants through the use of the SSRC and CSRC fields. In 1759 addition, it is capable of carrying some further identity information 1760 about these participants using the RTCP Source Descriptors (SDES). 1761 In topologies that provide a full all-to-all functionality, i.e. 1762 ASM, Mesh with common RTP session, Relay a compliant RTP 1763 implementation offers the functionality directly as specified in RTP. 1764 In topologies that do not offer all-to-all communication, it is 1765 necessary that RTCP is handled correctly in domain bridging function. 1766 RTP includes explicit specification text for Translators and Mixers, 1767 and for SFMs the required functionality can be derived from that 1768 text. However, the MCU described in Section 3.8 cannot offer the 1769 full functionality for session participant identification through RTP 1770 means. The topologies that create independent RTP sessions per End 1771 Point or pair of End Points, like Back-to-Back RTP session, MESH with 1772 independent RTP sessions, and the RTCP terminating MCU RTCP 1773 terminating MCU (Section 3.9) do not support RTP based identification 1774 of session participants. In all those cases, other non-RTP based 1775 mechanisms need to be implemented if such knowledge is required or 1776 desirable. 1778 4.1.6. Loop Detection 1780 In complex topologies with multiple interconnected domains, it is 1781 possible to unintentionally form media loops. RTP and RTCP support 1782 detecting such loops, as long as the SSRC and CSRC identities are 1783 maintained and correctly set in forwarded packets. Loop detection 1784 will work in ASM, SSM, Mesh with joint RTP session, and Relay. It is 1785 likely that loop detection works for the video switching MCU 1786 Section 3.8, at least as long as it forwards the RTCP between the End 1787 Points. However, the Back-to-Back RTP sessions, Mesh with 1788 independent RTP sessions, SFM, will definitely break the loop 1789 detection mechanism. 1791 4.2. Comparison of Topologies 1793 The table below attempts to summarize the properties of the different 1794 topologies. The legend to the topology abbreviations are: Topo- 1795 Point-to-Point (PtP), Topo-ASM (ASM), Topo-SSM (SSM), Topo-Trns- 1796 Translator (TT), Topo-Media-Translator (including Transport 1797 Translator) (MT), Topo-Mesh with joint session (MJS), Topo-Mesh with 1798 individual sessions (MIS), Topo-Mixer (Mix), Topo-Asymmetric (ASY), 1799 Topo-Video-switch-MCU (VSM), and Topo-RTCP-terminating-MCU (RTM), 1800 Selective Forwarding Middlebox (SFM). In the table below, Y 1801 indicates Yes or full support, N indicates No support, (Y) indicates 1802 partial support, and N/A indicates not applicable. 1804 Property PtP ASM SSM TT MT MJS MIS Mix ASY VSM RTM SFM 1805 --------------------------------------------------------------------- 1806 All to All media N Y (Y) Y Y Y (Y) (Y) (Y) (Y) (Y) (Y) 1807 Interoperability N/A N N Y Y Y Y Y Y N Y Y 1808 Per Domain Adaptation N/A N N N Y N Y Y Y N Y Y 1809 Aggregation of media N N N N N N N Y (Y) Y Y N 1810 Full Session View Y Y Y Y Y Y N Y Y (Y) N Y 1811 Loop Detection Y Y Y Y Y Y N Y Y (Y) N N 1813 Please note that the Media Translator also includes the transport 1814 Translator functionality. 1816 5. Security Considerations 1818 The use of Mixers, SFMs and Translators has impact on security and 1819 the security functions used. The primary issue is that both Mixers, 1820 SFMs and Translators modify packets, thus preventing the use of 1821 integrity and source authentication, unless they are trusted devices 1822 that take part in the security context, e.g., the device can send 1823 Secure Realtime Transport Protocol (SRTP) and Secure Realtime 1824 Transport Control Protocol (SRTCP) [RFC3711] packets to End Points in 1825 the Communication Session. If encryption is employed, the media 1826 Translator, SFM and Mixer need to be able to decrypt the media to 1827 perform its function. A transport Translator may be used without 1828 access to the encrypted payload in cases where it translates parts 1829 that are not included in the encryption and integrity protection, for 1830 example, IP address and UDP port numbers in a media stream using SRTP 1831 [RFC3711]. However, in general, the Translator, SFM or Mixer needs 1832 to be part of the signalling context and get the necessary security 1833 associations (e.g., SRTP crypto contexts) established with its RTP 1834 session participants. 1836 Including the Mixer, SFM and Translator in the security context 1837 allows the entity, if subverted or misbehaving, to perform a number 1838 of very serious attacks as it has full access. It can perform all 1839 the attacks possible (see RFC 3550 and any applicable profiles) as if 1840 the media session were not protected at all, while giving the 1841 impression to the human session participants that they are protected. 1843 Transport Translators have no interactions with cryptography that 1844 works above the transport layer, such as SRTP, since that sort of 1845 Translator leaves the RTP header and payload unaltered. Media 1846 Translators, on the other hand, have strong interactions with 1847 cryptography, since they alter the RTP payload. A media Translator 1848 in a session that uses cryptographic protection needs to perform 1849 cryptographic processing to both inbound and outbound packets. 1851 A media Translator may need to use different cryptographic keys for 1852 the inbound and outbound processing. For SRTP, different keys are 1853 required, because an RFC 3550 media Translator leaves the SSRC 1854 unchanged during its packet processing, and SRTP key sharing is only 1855 allowed when distinct SSRCs can be used to protect distinct packet 1856 streams. 1858 When the media Translator uses different keys to process inbound and 1859 outbound packets, each session participant needs to be provided with 1860 the appropriate key, depending on whether they are listening to the 1861 Translator or the original source. (Note that there is an 1862 architectural difference between RTP media translation, in which 1863 participants can rely on the RTP Payload Type field of a packet to 1864 determine appropriate processing, and cryptographically protected 1865 media translation, in which participants must use information that is 1866 not carried in the packet.) 1868 When using security mechanisms with Translators, SFMs and Mixers, it 1869 is possible that the Translator, SFM or Mixer could create different 1870 security associations for the different domains they are working in. 1871 Doing so has some implications: 1873 First, it might weaken security if the Mixer/Translator accepts a 1874 weaker algorithm or key in one domain than in another. Therefore, 1875 care should be taken that appropriately strong security parameters 1876 are negotiated in all domains. In many cases, "appropriate" 1877 translates to "similar" strength. If a key management system does 1878 allow the negotiation of security parameters resulting in a different 1879 strength of the security, then this system should notify the 1880 participants in the other domains about this. 1882 Second, the number of crypto contexts (keys and security related 1883 state) needed (for example, in SRTP [RFC3711]) may vary between 1884 Mixers, SFMs and Translators. A Mixer normally needs to represent 1885 only a single SSRCs per domain and therefore needs to create only one 1886 security association (SRTP crypto context) per domain. In contrast, 1887 a Translator needs one security association per participant it 1888 translates towards, in the opposite domain. Considering Figure 11, 1889 the Translator needs two security associations towards the multicast 1890 domain, one for B and one for D. It may be forced to maintain a set 1891 of totally independent security associations between itself and B and 1892 D respectively, so as to avoid two-time pad occurrences. These 1893 contexts must also be capable of handling all the sources present in 1894 the other domains. Hence, using completely independent security 1895 associations (for certain keying mechanisms) may force a Translator 1896 to handle N*DM keys and related state; where N is the total number of 1897 SSRCs used over all domains and DM is the total number of domains. 1899 The multicast based (ASM and SSM), Relay and Mesh with common RTP 1900 session are all topologies with multiple End Points that require 1901 shared knowledge about the different crypto contexts for the End 1902 Points. These multi-party topologies have special requirements on 1903 the key-management as well as the security functions. Specifically 1904 source-authentication in these environments has special requirements. 1906 There exist a number of different mechanisms to provide keys to the 1907 different participants. One example is the choice between group keys 1908 and unique keys per SSRC. The appropriate keying model is impacted 1909 by the topologies one intends to use. The final security properties 1910 are dependent on both the topologies in use and the keying 1911 mechanisms' properties, and need to be considered by the application. 1912 Exactly which mechanisms are used is outside of the scope of this 1913 document. Please review RTP Security Options [RFC7201] to get a 1914 better understanding of most of the available options. 1916 6. IANA Considerations 1918 This document makes no request of IANA. 1920 Note to RFC Editor: this section may be removed on publication as an 1921 RFC. 1923 7. Acknowledgements 1925 The authors would like to thank Bo Burman, Umesh Chandra, Roni Even, 1926 Keith Lantz, Ladan Gharai, Geoff Hunt, Mark Baugher, and Alex 1927 Eleftheriadis for their help in reviewing this document. 1929 8. References 1931 8.1. Normative References 1933 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 1934 Jacobson, "RTP: A Transport Protocol for Real-Time 1935 Applications", STD 64, RFC 3550, July 2003. 1937 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 1938 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 1939 RFC 3711, March 2004. 1941 [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session 1942 Initiation Protocol (SIP) Event Package for Conference 1943 State", RFC 4575, August 2006. 1945 [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, 1946 "Extended RTP Profile for Real-time Transport Control 1947 Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July 1948 2006. 1950 8.2. Informative References 1952 [I-D.ietf-avtcore-rtp-multi-stream-optimisation] 1953 Lennox, J., Westerlund, M., Wu, W., and C. Perkins, 1954 "Sending Multiple Media Streams in a Single RTP Session: 1955 Grouping RTCP Reception Statistics and Other Feedback", 1956 draft-ietf-avtcore-rtp-multi-stream-optimisation-03 (work 1957 in progress), July 2014. 1959 [RFC1112] Deering, S., "Host extensions for IP multicasting", STD 5, 1960 RFC 1112, August 1989. 1962 [RFC3022] Srisuresh, P. and K. Egevang, "Traditional IP Network 1963 Address Translator (Traditional NAT)", RFC 3022, January 1964 2001. 1966 [RFC3569] Bhattacharyya, S., "An Overview of Source-Specific 1967 Multicast (SSM)", RFC 3569, July 2003. 1969 [RFC4607] Holbrook, H. and B. Cain, "Source-Specific Multicast for 1970 IP", RFC 4607, August 2006. 1972 [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, 1973 "Codec Control Messages in the RTP Audio-Visual Profile 1974 with Feedback (AVPF)", RFC 5104, February 2008. 1976 [RFC5760] Ott, J., Chesterfield, J., and E. Schooler, "RTP Control 1977 Protocol (RTCP) Extensions for Single-Source Multicast 1978 Sessions with Unicast Feedback", RFC 5760, February 2010. 1980 [RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using 1981 Relays around NAT (TURN): Relay Extensions to Session 1982 Traversal Utilities for NAT (STUN)", RFC 5766, April 2010. 1984 [RFC6465] Ivov, E., Marocco, E., and J. Lennox, "A Real-time 1985 Transport Protocol (RTP) Header Extension for Mixer-to- 1986 Client Audio Level Indication", RFC 6465, December 2011. 1988 [RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP 1989 Sessions", RFC 7201, April 2014. 1991 Authors' Addresses 1993 Magnus Westerlund 1994 Ericsson 1995 Farogatan 6 1996 SE-164 80 Kista 1997 Sweden 1999 Phone: +46 10 714 82 87 2000 Email: magnus.westerlund@ericsson.com 2002 Stephan Wenger 2003 Vidyo 2004 433 Hackensack Ave 2005 Hackensack, NJ 07601 2006 USA 2008 Email: stewe@stewe.org